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You may not use this document on any fee based training course
without prior consent from the author.

The Little Asterisk Handbook is dedicated to
my wife Jennie, without who, I'd have neither the inclination or time to write
it.

Document Conventions

This document uses
some simple conventions to represent how the information contained should be
viewed by the reader.

Elements that
appear in boxes like the one below are designed to be typed in to the
asterisk/linux machine directly by the user

ls -lah

Responses from the
machine or screen display output appear in boxes such as the one below,

total 832K

drwxr-xr-x8 rootroot3.0K
Apr 15 15:36 .

drwxr-xr-x63 rootroot5.0K
Apr 14 13:28 ..

-rw-r--r--1 rootroot1.8K
Dec42003 2q

-rwxr-xr-x1 rootroot140
Dec42003 adsi.conf

-rwxr-xr-x1 rootroot840
Dec42003 adtranvofr.conf

The output may be
trimmed or edited for brevity. If there is a particular item the reader should
take note of it may appear as:

You
may find notes indicated by the symbol to the left useful or time saving.
Notes are in italics to separate from the text body.

Introduction

You probably
already know this but it is worth reinforcing the idea. Asterisk tears the arms
and legs off traditional PBX systems, and beats them with the soggy ends.

Traditional PBX's
allow you to perform the functions that the manufacturer thinks you might want
or need. Asterisk allows you to do this and more. You get full control and the
ability to extend and expand its functionality – in reality, your imagination
is the limit.

What? You don't
believe me? How about this as an example. I've left the house and forgotten to
take my keys.. oh no, how am I going to get back in? Simple. I have a special
number I can call, enter my passwords and have the door catch release – all
controlled from my PBX.

Do not use
Caller ID as the sole method of authenticating callers, in reality caller id
is simply an indication of who might be calling and is easily
spoofed. Use Caller ID in conjunction with passwords for items requiring
security.

Asterisk can be
easily expanded by adding your own modules or, remember we're dealing with open
source here, you can change the existing modules to suit your specific needs.

Commercial PBX's
offering only a subset of the features in Asterisk retail for huge sums of
money, be thankful that Digium made their code GPL so you, me and your pet
dog/cat/goldfish/badger could enjoy the power of Asterisk for free. It is my
firm belief that Asterisk has made a huge contribution to VoIP and its
acceptance.

One thing I will
not mention in this document is Skype – well ok, I'll mention it in passing but
in my opinion it is bad for VoIP simply because it's a closed system. For me
the freedom I get from VoIP systems and what makes them powerful is their
ability to inter operate with other systems using STANDARDS. In the
world of Skype, if you aren't using their client, you don't exist. Do yourself
a favor, forget about Skype and join something like Free World Dialup ( http://www.freeworlddialup.com ) - It's free and inter operates with most
networks.

Please note that
this document is in constant flux, as a result things may change or you may
find yourself drifting through space and time. I accept no responsibility for
this. Please ensure you have adequate insurance to cover any such eventuality.
No return keys were harmed in the making of this document, any references to
'hit the return key' should not be taken literally, hammers should not be used.

Target Audience

I'll confess
straight up, this document is designed for people who have Asterisk installed
already or are using something like my Asterisk Live! CD, CF card or PXE
editions found at my website http://www.automated.it/asterisk/ . It will not deal with installing
Asterisk, there are numerous guides available, including my own Getting
StartedWith Asterisk Guide at http://www.automated.it/guidetoasterisk.htm – Nothing like a bit of self promotion, eh?
This Document is intended to help you configure an installed Asterisk System

There are other
versions of Live installs of asterisk and I've listed some at the end of this
document in the useful links section.

The document is
split into three different sections, the basics, a simple configuration and
advanced and custom setup. The basics will introduce the main concepts and
explain the essential information you need in order to understand how the
configuration elements hang together.

Special
Thanks

Special thanks go to Ian Lowe of Wintermute Ltd ( http://www.wintermute-ltd.com ) and James Hoye,for proof reading this document.

What is Asterisk?

Asterisk PBX, from
now on just called Asterisk, is Linux based, Open Source and free PBX software. Or to quote
from the Asterisk website http://www.asterisk.org (with corrected spelling
;) )

Asterisk is a
complete PBX in software. It runs on Linux and provides all of the features you would expect
from a PBX and more. Asterisk does voice over IP in three protocols, and can inter
operate with almost all standards-based telephony equipment using comparatively
inexpensive hardware.

Asterisk provides
Voice mail services with Directory, Call Conferencing, Interactive Voice Response,
Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323
(as both client and gateway). Check the Features section for a more complete list.

Asterisk needs no
additional hardware for Voice over IP. For interconnection with digital and analog
telephony equipment, Asterisk supports a number of hardware devices, most notably all of the
hardware manufactured by Asterisk's sponsors, Digium. Digium has single and quad
span T1 and E1 interfaces for interconnection to PRI lines and channel banks. In addition, an analog FXO card is
available, and more analog interfaces are in the works.

Who Made Asterisk?

Asterisk was
originally written by Mark Spencer of Digium dba Linux Support Services, Inc. Code has been
contributed from Open Source coders around the world, and testing and bug-patches
from the community have provided invaluable aid to the development of this
software.

The Basics

With the exception
of one file (/etc/zaptel.conf) all of the Asterisk configuration files live in /etc/asterisk/
. Listing them produces a somewhat daunting array of about 40 files. The good
news is that you don't have to edit all of them, you'll only need to modify
those that you need or want to use.

The single most
important file you need to understand is extensions.conf, this file defines the
extensions (phone numbers) and actions carried out when an extension is called.
It is the core element of Asterisk configuration, everything else you do relies
on this.

Of course this
creates a little problem for someone starting out. To use and understand the
extensions.conf file, you need to use and understand the configuration files
that define phones so that you can test as see for yourself how things hang
together. As a consequence of this the first thing we'll do is setup a phone
blindly for use in our testing, then we'll go into detail on the extensions
file. When I say blindly, I simply mean that I'll tell you what to type and
you'll type it without particularly understanding it. Don't worry we'll come
back to it later and I'll explain everything.

Setup A Test Phone

For our test phone
we're going to use SIP, and it's going to be a softphone. Some of you may be
squealing like little children at this point, complaining that you've just
spent your hard earned money on a hardphone and you want to use that. No
problem, you can use that, but since there are so many makes and models YOU
will need to configure the phone side on your own.

A softphone
is simply a piece of software for your PC that looks a bit like a phone, a
hardphone is something solid, made of plastic/metal which will need a power
supply and network connection it resembles a real telephone.

Why SIP? Well,
there's a large array of usable soft and hard phones out there that all use SIP
which means it's easy for you to find a free one or one you are comfortable
with. The first thing you're going to need to do is download and install a
softphone. For the remainder of this document I'm going to assume that you've
downloaded and installed Xlite from Xten software. Xlite can be downloaded
fromthe Xten website, http://www.xten.com/index.php?menu=products&smenu=download .

You will, at the
minimum, need a sound card and some speakers in your PC/Mac, that way you'll at
least get to hear things when we demonstrate them, you really should have a
microphone too but if it's not a headset then be very aware of potential
feedback.

The first thing
we'll do is setup the Asterisk side, then we'll go on to configure the phone.
Connect to your Asterisk machine via either the console or an application like
Putty ( http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html ) and login as the root user.Since you've already learned that Asterisk
configuration files live in /etc/asterisk (ahem, ok a little reminder never
hurt) we should go to that directory, type the following in the console/putty
window and hit return

cd
/etc/asterisk

Since we're dealing
with SIP, we need to modify the sip configuration file, or sip.conf (that was
difficult eh?!)

vi sip.conf

vi is an editor for
linux machines, more than likely it's really vim not vi on your machine, but
let's not dwell on that. When you first start up vi you'll be in command
mode.

Here are a
few tips for using vi:

Hit the
insert key once to get into insert mode

Hit it again
to get into replace mode

Hit the
escape key to get into command mode

To quit
without saving, hit the escape key followed by a colon (:) then lower case Q
and an exclamation mark (!) then hit return,eg:

:q!

To save a
file and then quit, hit the escape key followed by a colon (:) then lower
case W and lower case Q

then hit
return eg:

:wq

Hint:

If you'd prefer
not to use vi, try nano or pico but remember the save/exit commands are
different!

We need to define
an account for this install, for the moment scroll all the way down to the end
of the file and add these lines:

[1000]

username=1000

secret=abc123

context=mytest

host=dynamic

Then save the file.

Now that you've
installed X-lite, start it up, you should see two windows, the main phone
interface window and the configuration window.

We're going to
configure the softphone, save the settings and then shut it down and continue
with the document, and since I haven't told you to start Asterisk yet, you
haven't – right? If you have, then connect to the server and stop Asterisk
using the commands

asterisk -r

And then issuing
the stop now command in the Asterisk cli.

stop now

You should see
something like

asterisk*CLI>
stop now

asterisk*CLI>
Asterisk ended with exit status 0

Asterisk
shutdown normally.

Disconnected
from Asterisk server

If you've behaved
yourself and not started Asterisk you get a gold star. In Xlite's configuration
window change the following to their corresponding value:

EnabledYes

Display NameYour Name

Username1000

Authorisation User1000

Passwordabc123

SIP ProxyThe IP address of your Asterisk server

Domain/RealmThe IP address of your Asterisk server (if you have an FQDN

(domain
name) then you can use that )

RegisterAlways

Scroll all the way
down to the bottom of the list and double click Update settings, and
then double click Confirm Update Settings. Close the system settings
window and the Xlite main window. Be careful, Xlite likes to 'close' to the
system tray. If you see a yellow box like this (the red ring is just to highlight
the icon, ok, don't complain yours doesn't have a red ring around it...
sheeesssh some people).

Xlite is still
running, right click on it and select Exit. Xlite is now ready for use with
Asterisk, but Asterisk isn't quite ready for Xlite just yet...